1,481 research outputs found
Noise adaptive training for subspace Gaussian mixture models
Noise adaptive training (NAT) is an effective approach to normalise the environmental distortions in the training data. This paper investigates the model-based NAT scheme using joint uncertainty decoding (JUD) for subspace Gaussian mixture models (SGMMs). A typical SGMM acoustic model has much larger number of surface Gaussian components, which makes it computationally infeasible to compensate each Gaussian explicitly. JUD tackles the problem by sharing the compensation parameters among the Gaussians and hence reduces the computational and memory demands. For noise adaptive training, JUD is reformulated into a generative model, which leads to an efficient expectation-maximisation (EM) based algorithm to update the SGMM acoustic model parameters. We evaluated the SGMMs with NAT on the Aurora 4 database, and obtained higher recognition accuracy compared to systems without adaptive training. Index Terms: adaptive training, noise robustness, joint uncertainty decoding, subspace Gaussian mixture model
DNN-based uncertainty estimation for weighted DNN-HMM ASR
In this paper, the uncertainty is defined as the mean square error between a
given enhanced noisy observation vector and the corresponding clean one. Then,
a DNN is trained by using enhanced noisy observation vectors as input and the
uncertainty as output with a training database. In testing, the DNN receives an
enhanced noisy observation vector and delivers the estimated uncertainty. This
uncertainty in employed in combination with a weighted DNN-HMM based speech
recognition system and compared with an existing estimation of the noise
cancelling uncertainty variance based on an additive noise model. Experiments
were carried out with Aurora-4 task. Results with clean, multi-noise and
multi-condition training are presented
Efficient Invariant Features for Sensor Variability Compensation in Speaker Recognition
In this paper, we investigate the use of invariant features for speaker recognition. Owing to their characteristics, these features are introduced to cope with the difficult and challenging problem of sensor variability and the source of performance degradation inherent in speaker recognition systems. Our experiments show: (1) the effectiveness of these features in match cases; (2) the benefit of combining these features with the mel frequency cepstral coefficients to exploit their discrimination power under uncontrolled conditions (mismatch cases). Consequently, the proposed invariant features result in a performance improvement as demonstrated by a reduction in the equal error rate and the minimum decision cost function compared to the GMM-UBM speaker recognition systems based on MFCC features
Speech Recognition Front End Without Information Loss
Speech representation and modelling in high-dimensional spaces of acoustic
waveforms, or a linear transformation thereof, is investigated with the aim of
improving the robustness of automatic speech recognition to additive noise. The
motivation behind this approach is twofold: (i) the information in acoustic
waveforms that is usually removed in the process of extracting low-dimensional
features might aid robust recognition by virtue of structured redundancy
analogous to channel coding, (ii) linear feature domains allow for exact noise
adaptation, as opposed to representations that involve non-linear processing
which makes noise adaptation challenging. Thus, we develop a generative
framework for phoneme modelling in high-dimensional linear feature domains, and
use it in phoneme classification and recognition tasks. Results show that
classification and recognition in this framework perform better than analogous
PLP and MFCC classifiers below 18 dB SNR. A combination of the high-dimensional
and MFCC features at the likelihood level performs uniformly better than either
of the individual representations across all noise levels
Spoofing Detection Goes Noisy: An Analysis of Synthetic Speech Detection in the Presence of Additive Noise
Automatic speaker verification (ASV) technology is recently finding its way
to end-user applications for secure access to personal data, smart services or
physical facilities. Similar to other biometric technologies, speaker
verification is vulnerable to spoofing attacks where an attacker masquerades as
a particular target speaker via impersonation, replay, text-to-speech (TTS) or
voice conversion (VC) techniques to gain illegitimate access to the system. We
focus on TTS and VC that represent the most flexible, high-end spoofing
attacks. Most of the prior studies on synthesized or converted speech detection
report their findings using high-quality clean recordings. Meanwhile, the
performance of spoofing detectors in the presence of additive noise, an
important consideration in practical ASV implementations, remains largely
unknown. To this end, we analyze the suitability of state-of-the-art synthetic
speech detectors under additive noise with a special focus on front-end
features. Our comparison includes eight acoustic feature sets, five related to
spectral magnitude and three to spectral phase information. Our extensive
experiments on ASVSpoof 2015 corpus reveal several important findings. Firstly,
all the countermeasures break down even at relatively high signal-to-noise
ratios (SNRs) and fail to generalize to noisy conditions. Secondly, speech
enhancement is not found helpful. Thirdly, GMM back-end generally outperforms
the more involved i-vector back-end. Fourthly, concerning the compared
features, the Mel-frequency cepstral coefficients (MFCCs) and subband spectral
centroid magnitude coefficients (SCMCs) perform the best on average though the
winner method depends on SNR and noise type. Finally, a study with two score
fusion strategies shows that combining different feature based systems improves
recognition accuracy for known and unknown attacks in both clean and noisy
conditions.Comment: 23 Pages, 7 figure
Adverse Conditions and ASR Techniques for Robust Speech User Interface
The main motivation for Automatic Speech Recognition (ASR) is efficient
interfaces to computers, and for the interfaces to be natural and truly useful,
it should provide coverage for a large group of users. The purpose of these
tasks is to further improve man-machine communication. ASR systems exhibit
unacceptable degradations in performance when the acoustical environments used
for training and testing the system are not the same. The goal of this research
is to increase the robustness of the speech recognition systems with respect to
changes in the environment. A system can be labeled as environment-independent
if the recognition accuracy for a new environment is the same or higher than
that obtained when the system is retrained for that environment. Attaining such
performance is the dream of the researchers. This paper elaborates some of the
difficulties with Automatic Speech Recognition (ASR). These difficulties are
classified into Speakers characteristics and environmental conditions, and
tried to suggest some techniques to compensate variations in speech signal.
This paper focuses on the robustness with respect to speakers variations and
changes in the acoustical environment. We discussed several different external
factors that change the environment and physiological differences that affect
the performance of a speech recognition system followed by techniques that are
helpful to design a robust ASR system.Comment: 10 pages 2 Table
On Single-Channel Speech Enhancement and On Non-Linear Modulation-Domain Kalman Filtering
This report focuses on algorithms that perform single-channel speech
enhancement. The author of this report uses modulation-domain Kalman filtering
algorithms for speech enhancement, i.e. noise suppression and dereverberation,
in [1], [2], [3], [4] and [5]. Modulation-domain Kalman filtering can be
applied for both noise and late reverberation suppression and in [2], [1], [3]
and [4], various model-based speech enhancement algorithms that perform
modulation-domain Kalman filtering are designed, implemented and tested. The
model-based enhancement algorithm in [2] estimates and tracks the speech phase.
The short-time-Fourier-transform-based enhancement algorithm in [5] uses the
active speech level estimator presented in [6]. This report describes how
different algorithms perform speech enhancement and the algorithms discussed in
this report are addressed to researchers interested in monaural speech
enhancement. The algorithms are composed of different processing blocks and
techniques [7]; understanding the implementation choices made during the system
design is important because this provides insights that can assist the
development of new algorithms. Index Terms - Speech enhancement,
dereverberation, denoising, Kalman filter, minimum mean squared error
estimation.Comment: 13 page
Studies on noise robust automatic speech recognition
Noise in everyday acoustic environments such as cars, traffic environments, and cafeterias remains one of the main challenges in automatic speech recognition (ASR). As a research theme, it has received wide attention in conferences and scientific journals focused on speech technology. This article collection reviews both the classic and novel approaches suggested for noise robust ASR. The articles are literature reviews written for the spring 2009 seminar course on noise robust automatic speech recognition (course code T-61.6060) held at TKK
Robust speech recognition based on a Bayesian prediction approach
We study a category of robust speech recognition problem in which mismatches exist between training and testing conditions, and no accurate knowledge of the mismatch mechanism is available. The only available information is the test data along with a set of pretrained Gaussian mixture continuous density hidden Markov models (CDHMMs). We investigate the problem from the viewpoint of Bayesian prediction. A simple prior distribution, namely constrained uniform distribution, is adopted to characterize the uncertainty of the mean vectors of the CDHMMs. Two methods, namely a model compensation technique based on Bayesian predictive density and a robust decision strategy called Viterbi Bayesian predictive classification are studied. The proposed methods are compared with the conventional Viterbi decoding algorithm in speaker-independent recognition experiments on isolated digits and TI connected digit strings (TIDTGITS), where the mismatches between training and testing conditions are caused by: (1) additive Gaussian white noise, (2) each of 25 types of actual additive ambient noises, and (3) gender difference. The experimental results show that the adopted prior distribution and the proposed techniques help to improve the performance robustness under the examined mismatch conditions.published_or_final_versio
ROBUST SPEAKER RECOGNITION BASED ON LATENT VARIABLE MODELS
Automatic speaker recognition in uncontrolled environments is a very challenging task due to channel distortions, additive noise and reverberation. To address these issues, this thesis studies probabilistic latent variable models of short-term spectral information that leverage large amounts of data to achieve robustness in challenging conditions.
Current speaker recognition systems represent an entire speech utterance as a single point in a high-dimensional space. This representation is known as "supervector". This thesis starts by analyzing the properties of this representation. A novel visualization procedure of supervectors is presented by which qualitative insight about the information being captured is obtained. We then propose the use of an overcomplete dictionary to explicitly decompose a supervector into a speaker-specific component and an undesired variability component. An algorithm to learn the dictionary from a large collection of data is discussed and analyzed. A subset of the entries of the dictionary is learned to represent speaker-specific information and another subset to represent distortions. After encoding the supervector as a linear combination of the dictionary entries, the undesired variability is removed by discarding the contribution of the distortion components. This paradigm is closely related to the previously proposed paradigm of Joint Factor Analysis modeling of supervectors. We establish a connection between the two approaches and show how our proposed method provides improvements in terms of computation and recognition accuracy.
An alternative way to handle undesired variability in supervector representations is to first project them into a lower dimensional space and then to model them in the reduced subspace. This low-dimensional projection is known as "i-vector". Unfortunately, i-vectors exhibit non-Gaussian behavior, and direct statistical modeling requires the use of heavy-tailed distributions for optimal performance. These approaches lack closed-form solutions, and therefore are hard to analyze. Moreover, they do not scale well to large datasets. Instead of directly modeling i-vectors, we propose to first apply a non-linear transformation and then use a linear-Gaussian model. We present two alternative transformations and show experimentally that the transformed i-vectors can be optimally modeled by a simple linear-Gaussian model (factor analysis). We evaluate our method on a benchmark dataset with a large amount of channel variability and show that the results compare favorably against the competitors. Also, our approach has closed-form solutions and scales gracefully to large datasets.
Finally, a multi-classifier architecture trained on a multicondition fashion is proposed to address the problem of speaker recognition in the presence of additive noise. A large number of experiments are conducted to analyze the proposed architecture and to obtain guidelines for optimal performance in noisy environments. Overall, it is shown that multicondition training of multi-classifier architectures not only produces great robustness in the anticipated conditions, but also generalizes well to unseen conditions
- …